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AgeCommit message (Expand)Author
2014-08-26Remove Android.mk build files.pbos@webrtc.org
2014-08-26Remove former team members from OWNERS and WATCHLISTSkjellander@webrtc.org
2014-08-11Remove more dependencies on openssl, add dependency on boringssl. Continues o...henrike@webrtc.org
2014-08-06RTP video playback tool using Call APIs.pbos@webrtc.org
2014-08-06Fix crashing fake network pipe tests.stefan@webrtc.org
2014-08-06Add end-to-end H.264 packetization test.stefan@webrtc.org
2014-07-31Add simulation of network effects to video_loopback tool.stefan@webrtc.org
2014-07-10Remove more unused tsan suppressions and fix call test passing the same decod...andresp@webrtc.org
2014-07-10Support VP8 encoder settings in VideoSendStream.pbos@webrtc.org
2014-07-09Add full stack test cases with a fake network pipe.stefan@webrtc.org
2014-07-08Some refactoring inside rtp_rtcp/.pbos@webrtc.org
2014-07-07Preserve RTP states for restarted VideoSendStreams.pbos@webrtc.org
2014-07-07Remove GetDefaultConfigs() from Call.pbos@webrtc.org
2014-07-04Add pbos@webrtc.org as owner for webrtc/test/.pbos@webrtc.org
2014-07-04Add boilerplate code for H.264.stefan@webrtc.org
2014-06-30Reserve RTP/RTCP modules in SetSSRC.pbos@webrtc.org
2014-06-30Removing W3C conformance tests after move to web-platform-tests.phoglund@webrtc.org
2014-06-27Refactor Call-based tests.pbos@webrtc.org
2014-06-25Add RTCP packet types to packet builder:asapersson@webrtc.org
2014-06-17Updated W3C getusermedia tests to the latest version of the spec.phoglund@webrtc.org
2014-06-16Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.asapersson@webrtc.org
2014-06-16Remove ivinnichenko from webrtc/test/OWNERSkjellander@webrtc.org
2014-06-13Pass GYP DEPTH variable to isolate.kjellander@webrtc.org
2014-06-10Add kjellander@webrtc.org as OWNER for *.isolatekjellander@webrtc.org
2014-06-09Updated conformance tests and w3c-ified them.phoglund@webrtc.org
2014-06-06Make VideoSendStream/VideoReceiveStream configs const.pbos@webrtc.org
2014-06-05Fix the chain that propagates the audio frame's rtp and ntp timestamp including:wu@webrtc.org
2014-06-05Adding back platform specific renderer to video loopback test.mflodman@webrtc.org
2014-06-04Android: cleanup gtest_target_type conditions.henrike@webrtc.org
2014-05-21Switch to using base/constructormagic.h and remove system_wrappers/interface/...henrike@webrtc.org
2014-05-21Revert 6202 "Switch to using base/constructormagic.h and remove ..."mcasas@webrtc.org
2014-05-20Switch to using base/constructormagic.h and remove system_wrappers/interface/...henrike@webrtc.org
2014-05-20Add NACK and RPSI packet types to RTCP packet builder.asapersson@webrtc.org
2014-05-19Add interface to propagate audio capture timestamp to the renderer.wu@webrtc.org
2014-05-16Wire up --force_fieldtrials for vie_auto_test and for test targets linking wi...andresp@webrtc.org
2014-05-14Add DeliveryStatus enum to DeliverPacket().pbos@webrtc.org
2014-05-14Add webrtc field trials API.andresp@webrtc.org
2014-05-13Move gflags usage to video_loopback.pbos@webrtc.org
2014-04-29Added include of assert.h for files calling assert but missing the include.henrike@webrtc.org
2014-04-28Add thread annotations to Call API.pbos@webrtc.org
2014-04-25Replace scoped_array<T> with scoped_ptr<T[]>.andrew@webrtc.org
2014-04-24Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...wu@webrtc.org
2014-04-16Remove use of tmpnam.kjellander@webrtc.org
2014-04-14Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.fischman@webrtc.org
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
2014-03-19Remove internal codecs from VideoSendStream.pbos@webrtc.org
2014-03-13Implement minimum transmit bitrate.pbos@webrtc.org
2014-03-12Remove platform-specific code from new-API tests.pbos@webrtc.org
2014-03-01Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed.fischman@webrtc.org
2014-02-26Add SetConfig method to FakeNetworkPipe and to DirectTransporthenrik.lundin@webrtc.org
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